Switch browser side audio capture path to use base time primitives.

media/audio/mac/audio_low_latency_input_mac.cc

Index: media/audio/mac/audio_low_latency_input_mac.cc
diff --git a/media/audio/mac/audio_low_latency_input_mac.cc b/media/audio/mac/audio_low_latency_input_mac.cc
index cddf4300d3d61bf449a1cf9ebcc9725dc813233d..f86f2a50e4f4ae4d3a66efc581dd0f5896ddb505 100644
--- a/media/audio/mac/audio_low_latency_input_mac.cc
+++ b/media/audio/mac/audio_low_latency_input_mac.cc
@@ -250,7 +250,6 @@ AUAudioInputStream::AUAudioInputStream(
       sink_(nullptr),
       audio_unit_(0),
       input_device_id_(audio_device_id),
-      hardware_latency_frames_(0),
       number_of_channels_in_frame_(0),
       fifo_(input_params.channels(),
             number_of_frames_,
@@ -517,7 +516,7 @@ bool AUAudioInputStream::Open() {
   }
 
   // The hardware latency is fixed and will not change during the call.
-  hardware_latency_frames_ = GetHardwareLatency();
+  hardware_latency_ = GetHardwareLatency();
 
   // The master channel is 0, Left and right are channels 1 and 2.
   // And the master channel is not counted in |number_of_channels_in_frame_|.
@@ -960,7 +959,8 @@ OSStatus AUAudioInputStream::Provide(UInt32 number_of_frames,
     number_of_frames_provided_ = number_of_frames;
 
   // Update the capture latency.
-  double capture_latency_frames = GetCaptureLatency(time_stamp);
+  base::TimeDelta capture_latency = GetCaptureLatency(time_stamp);
+  base::TimeDelta delay_timestamp = base::TimeTicks::Now();
 
   // The AGC volume level is updated once every second on a separate thread.
   // Note that, |volume| is also updated each time SetVolume() is called
@@ -970,8 +970,6 @@ OSStatus AUAudioInputStream::Provide(UInt32 number_of_frames,
 
   AudioBuffer& buffer = io_data->mBuffers[0];
   uint8_t* audio_data = reinterpret_cast<uint8_t*>(buffer.mData);
-  uint32_t capture_delay_bytes = static_cast<uint32_t>(
-      (capture_latency_frames + 0.5) * format_.mBytesPerFrame);
   DCHECK(audio_data);
   if (!audio_data)
     return kAudioUnitErr_InvalidElement;
@@ -1003,7 +1001,8 @@ OSStatus AUAudioInputStream::Provide(UInt32 number_of_frames,
 
     // Compensate the audio delay caused by the FIFO.
     capture_delay_bytes += fifo_.GetAvailableFrames() * format_.mBytesPerFrame;

[o1ka, 2017/02/10 13:28:50]

Hmm.. does it really compile? :)

[DaleCurtis, 2017/02/11 01:43:13]

Like I said, not done yet :)

-    sink_->OnData(this, audio_bus, capture_delay_bytes, normalized_volume);
+    sink_->OnData(this, audio_bus, capture_latency, delay_timestamp,

[o1ka, 2017/02/10 13:28:50]

Shouldn't we adjust latency and timestamp values taking into account the fifo?

+                  normalized_volume);
   }
 
   return noErr;
@@ -1113,10 +1112,10 @@ int AUAudioInputStream::HardwareSampleRate() {
   return static_cast<int>(nominal_sample_rate);
 }
 
-double AUAudioInputStream::GetHardwareLatency() {
+base::TimeDelta AUAudioInputStream::GetHardwareLatency() {
   if (!audio_unit_ || input_device_id_ == kAudioObjectUnknown) {
     DLOG(WARNING) << "Audio unit object is NULL or device ID is unknown";
-    return 0.0;
+    return base::TimeDelta();
   }
 
   // Get audio unit latency.
@@ -1138,23 +1137,24 @@ double AUAudioInputStream::GetHardwareLatency() {
                                       nullptr, &size, &device_latency_frames);
   DLOG_IF(WARNING, result != noErr) << "Could not get audio device latency.";
 
-  return static_cast<double>((audio_unit_latency_sec * format_.mSampleRate) +
-                             device_latency_frames);
+  return base::TimeDelta::FromSecondsD(
+      audio_unit_latency_sec +
+      device_latency_frames / static_cast<double>(format_.mSampleRate));
 }
 
-double AUAudioInputStream::GetCaptureLatency(
+base::TimeDelta AUAudioInputStream::GetCaptureLatency(
     const AudioTimeStamp* input_time_stamp) {
   // Get the delay between between the actual recording instant and the time
   // when the data packet is provided as a callback.
   UInt64 capture_time_ns =
       AudioConvertHostTimeToNanos(input_time_stamp->mHostTime);
   UInt64 now_ns = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
-  double delay_frames = static_cast<double>(1e-9 * (now_ns - capture_time_ns) *
-                                            format_.mSampleRate);
 
   // Total latency is composed by the dynamic latency and the fixed
   // hardware latency.
-  return (delay_frames + hardware_latency_frames_);
+  return hardware_latency_ +
+         base::TimeDelta::FromMicroseconds((now_ns - capture_time_ns) /
+                                           kNanosecondsPerMicrosecond);
 }
 
 int AUAudioInputStream::GetNumberOfChannelsFromStream() {